Auditory Front End Customization

ABSTRACT

A method and system for implementing an acoustical front end customization are disclosed. The customization is implemented to optimize the sound level for each individual cochlear implant user. A known audio signal is generated using a sound source and captured by a microphone system. The captured sound signal is sampled at one or more locations along the signal processing pathway, and a spectrum is determined for the sampled signal and the known signal. A ratio of the two spectrums is related to the undesired transformation of the sampled signal, and a digital filter is designed based on the ratio to filter out the undesired transformation.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of and claims priority to theco-pending U.S. patent application Ser. No. 11/535,004, filed Sep. 25,2006, and entitled “Auditory Front End Customization,” which is herebyincorporated by reference.

TECHNICAL FIELD

The present disclosure relates to implantable neurostimulator devicesand systems, for example, cochlear stimulation systems, and to soundprocessing strategies employed in conjunction with such systems.

BACKGROUND

Prior to the past several decades, scientists generally believed that itwas impossible to restore hearing to the profoundly deaf. However,scientists have had increasing success in restoring normal hearing tothe deaf through electrical stimulation of the auditory nerve. Theinitial attempts to restore hearing were not very successful, aspatients were unable to understand speech. However, as scientistsdeveloped different techniques for delivering electrical stimuli to theauditory nerve, the auditory sensations elicited by electricalstimulation gradually came closer to sounding more like normal speech.The electrical stimulation is implemented through a prosthetic device,known as a cochlear implant (CI), which is implanted in the inner ear torestore partial hearing.

Cochlear implants generally employ an electrode array that is insertedinto the cochlear duct. One or more electrodes of the array selectivelystimulate the auditory nerve along different places in the cochlea basedon the frequency of a received acoustic signal picked up by a microphoneand transformed to an electrical signal by a digital signal processor(DSP) unit located in the external ear piece of a cochlear implant frontend.

After a patient has been provided with a CI, it is necessary toinitially “fit” or “adjust” the device during a fitting session. As usedherein, it should be noted that terms “fit”, “adjust”, “fitting”,“adjusting”, “program”, or “programming” relate to making electronic orsoftware programming changes to the CI device. A proper fitting isessential to ensuring the CI user experience natural sound quality.Currently, the fitting session suffers from inefficiency andsubjectivity for a few reasons. Because a new CI user is used toexperiencing either poor sound quality or no sound at all, he/she findsit difficult to qualitatively communicate a perceived sound quality andpreference to a technician during the fitting session. This results in afitted device not accurately tailored to the specific CI user. Worstyet, younger CI users (i.e. children) are incapable of communicatingeffectively the nature of experienced sound quality to the technician.

Characteristics of a cochlear implant front's end play an important rolein the perceived sound quality (and hence speech recognition or musicappreciation) experienced by the CI user. These characteristics aregoverned by the components of the front-end comprising a microphone, anND converter, and the acoustic effects resulting from a location of themicrophone on the user's head. While the component characteristics meetpre-defined standards, and can hence be compensated for, the acousticcharacteristics are unique to the CI user's anatomy and his/herplacement of the microphone on their head. Specifically, the uniqueshaping of the user's ears and head geometry can result in substantialshaping of the acoustic waveform picked up by the microphone. Becausethis shaping is unique to the CI user and his/her microphone placement,it cannot be compensated for with a generalized solution. This issue canbe even more critical in beamforming applications where signals frommultiple microphones are combined to achieve a desired directivity. Itis critical for the multiple microphones in these applications to havematched responses. Any differences in the microphones' responses due totheir placement on the patient's head can make this challenging.

SUMMARY

The methods and systems described here implement techniques foroptimizing sound levels as perceived through a cochlear implant. Forexample, the techniques are implemented to customize an acoustical frontend for each individual cochlear implant user.

In one aspect, a known audio signal is generated using a sound sourceand the generated audio signal is captured by a microphone system. Thecaptured sound signal is processed along one or more signal paths, andthe processed signal is sampled at one or more locations along thesignal processing pathway. Comparisons are made between the generatedknown audio signal and the sampled signal to determine the undesiredtransformation of the sampled signal. Based on the comparisons, adigital filter is designed to filter out the undesired transformationand customizing the acoustical front end for the individual cochlearimplant user.

Implementations can include one or more of the following features. Forexample, the known audio signal can be generated through an externalsound source. Also, the captured audio signal can be processed byconverting the known audio signal into an analog electrical signal;converting the analog electrical signal into a digital signal; andadjusting a gain of the digital signal. In addition, processed audiosignal can be sampled before adjusting the gain of the digital signal orafter adjusting the gain of the digital signal or both. Further, thesampled audio signal can be compared against the known audio signal bygenerating a spectrum of the sampled audio signal; generating a spectrumof the known audio signal; and determining a ratio of the sampled audiosignal spectrum over the known audio signal spectrum. Based on thedetermined ratio, the filter can be generated.

Implementations can also include one or more of the following features.For example, the processed signal can be sampled at two or morelocations along the signal path. In addition, the captured audio signalcan be processed along two or more signal paths in parallel. Further,the filter can be generated to optimally match a first response of afirst microphone with a second response of a second microphone. Thefirst and second responses of the first and second microphones comprisesa beamformer to provide directivity of the captured signal.

The techniques described in this specification can be implemented torealize one or more of the following advantages. For example, thetechniques can be implemented to eliminate the need to physically matchthe microphones at the manufacturing stage. The techniques can also beimplemented to provide a tailored fitting for individual CI user. Thetechniques also can be implemented to eliminate unknown loading effectsdue to the positioning of the two microphones and physiology of thepatient's head. The unknown loading effects created on the microphonesresults in unmatched responses even if the two microphones of a beamforming device are perfectly matched. Further, the techniques can beimplemented to compensate for the physiological effect. In addition, amicrophone test mode allows the user to run active or dynamic check/testof the microphone.

These general and specific aspects can be implemented using anapparatus, a method, a system, or any combination of an apparatus,methods, and systems. The details of one or more implementations are setforth in the accompanying drawings and the description below. Furtherfeatures, aspects, and advantages will become apparent from thedescription, the drawings, and the claims.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a functional block diagram of an auditory front endcustomization system.

FIG. 2 is a functional block diagram showing a detailed view of a soundprocessing portion.

FIG. 3A is a functional block diagram describing a single signal path.

FIG. 3B is a functional block diagram describing two signal paths.

FIG. 4 is a functional block diagram of the auditory front endcustomization system with the possible signal sampling locationsidentified.

FIG. 5 is a flowchart of a process of customizing the auditory frontend.

FIG. 6 presents a functional block diagram of an auditory front endcustomization system in a beamforming application.

FIG. 7 is a flowchart of a generalized method of customizing theauditory front end.

Like reference symbols indicate like elements throughout thespecification and drawings.

DETAILED DESCRIPTION

FIG. 1 depicts an auditory front end customization system 10 comprisinga fitting portion 100 in communication with a sound processing portion200. The fitting portion 100 can include a fitting system 110communicatively linked to an external sound source 130 using anappropriate communication link 120. The fitting system 110 may besubstantially as shown and described in U.S. Pat. Nos. 5,626,629 and6,289,247, both patents incorporated herein by reference. In general,the fitting system 110 is implemented on a computer system located at anoffice of an audiologist or a medical personnel and used to perform aninitial fitting or customization of a cochlear implant for a particularuser. The sound processing portion 200 is implemented on a behind theear (BTE) headpiece, which is shown and described in U.S. Pat. No.5,824,022, the patent incorporated herein by reference. The soundprocessing portion can include a microphone system 210 communicativelylinked to a sound processing system 230 using a suitable communicationlink 220. The fitting system 110 is coupled to the sound processingsystem 230 through an interface unit (IU) 140, or an equivalent device.A suitable communication link 150 couples the interface unit 140 withthe sound processing system 230 and the fitting system 110. The IU canbe included within the computer as a built-in I/O port including but notlimited to an IR port, serial port, a parallel port, and a USB port.

The fitting portion 100 can generate an acoustic signal, which can bepicked up and processed by the sound processing portion 200. Theprocessed acoustic signal can be passed to an implantable cochlearstimulator (ICS) 160 through an appropriate communication link 240. TheICS 160 is coupled to an electrode array 170 configured to be insertedwithin the cochlea of a patient. The implantable cochlear stimulator 170can apply the processed acoustic signal as a plurality of stimulatinginputs to a plurality of electrodes distributed along the electrodearray 170. The electrode array 170 may be substantially as shown anddescribed in U.S. Pat. Nos. 4,819,647 and 6,129,753, both patentsincorporated herein by reference. The sound processing portion 200 maybe substantially as shown and described in the co-pending U.S. patentapplication Ser. No. 11/003,155.

In some implementations, both the fitting portion 100 and the soundprocessing portion 200 are implemented in the external BTE headpiece.The fitting portion 100 can be controlled by a hand-held wired orwireless remote controller device (not shown) by the medical personnelor the cochlear implant user. The implantable cochlear stimulator 160and the electrode array 170 can be an internal, or implanted portion.Thus, a communication link 240 coupling the sound processing system 230and the internal portion can be a transcutaneous (through the skin) linkthat allows power and control signals to be sent from the soundprocessing system 230 to the implantable cochlear stimulator 160.

In some implementations, the sound processing portion 200 may beincorporated into an internally located implantable cochlear system (notshown).

The implantable cochlear stimulator can send information, such as dataand status signals, to the sound processing system 230 over thecommunication link 240. In order to facilitate bidirectionalcommunication between the sound processing system 230 and theimplantable cochlear stimulator 160, the communication link 240 caninclude more than one channel. Additionally, interference can be reducedby transmitting information on a first channel using anamplitude-modulated carrier and transmitting information on a secondchannel using a frequency-modulated carrier.

The communication links 120 and 220 are wired links using standard dataports such as Universal Serial Bus interface, IEEE 1394 FireWire, orother suitable serial or parallel port connections.

In some implementations, the communication links 120 and 220 arewireless links such as the Bluetooth protocol. The Bluetooth protocol isa short-range, low-power 1 Mbit/sec wireless network technology operatedin the 2.4 GHz band, which is appropriate for use in piconets. A piconetcan have a master and up to seven slaves. The master transmits in eventime slots, while slaves transmits in odd time slots. The devices in apiconet share a common communication data channel with total capacity of1 Mbit/sec. Headers and handshaking information are used by Bluetoothdevices to strike up a conversation and find each other to connect.

Other standard wireless links such as infrared, wireless fidelity(Wi-Fi), or any other suitable wireless connections can be implemented.Wi-Fi refers to any type of IEEE 802.11 protocol including802.11a/b/g/n. Wi-Fi generally provides wireless connectivity for adevice to the Internet or connectivity between devices. Wi-Fi operatesin the unlicensed 2.4 GHz radio bands, with an 11 Mbit/sec (802.11b) or54 Mbit/sec (802.11a) data rate or with products that contain bothbands. Infrared refers to light waves of a lower frequency out of rangeof what a human eye can perceive. Used in most television remote controlsystems, information is carried between devices via beams of infraredlight. The standard infrared system is called infrared data association(IrDA) and is used to connect some computers with peripheral devices indigital mode.

In implementations whereby the implantable cochlear stimulator 160 andthe electrode array 170 are implanted within the CI user, and themicrophone system 210 and the sound processing system 230 are carriedexternally (not implanted) by the CI user, the communication link 240can be realized through use of an antenna coil in the implantablecochlear stimulator and an external antenna coil coupled to the soundprocessing system 230. The external antenna coil can be positioned to bein alignment with the implantable cochlear stimulator, allowing thecoils to be inductively coupled to each other and thereby permittingpower and information, e.g., the stimulation signal, to be transmittedfrom the sound processing system 230 to the implantable cochlearstimulator 160.

In some implementations, the sound processing system 230 and theimplantable cochlear stimulator 160 are both be implanted within the CIuser, and the communication link 240 can be a direct-wired connection orother suitable link as shown in U.S. Pat. No. 6,308,101, incorporatedherein by reference.

FIG. 2 depicts major subsystems of the fitting system 110. In oneimplementation, the fitting system 110 includes a fitting software 38executable on a computer system 40 such as a personal computer, aportable computer, a mobile device, or other equivalent devices. Thecomputer system 40, with or without the IU 140, generates input signalsto the sound processing system 230 that stimulate acoustical signalsdetected by the microphone system 210. Depending on the situation, inputsignals generated by the computer system 40 can replace acoustic signalsnormally detected by the microphone system 210 or provide commandsignals that supplement the acoustic signals detected through themicrophone system 210. The fitting software 38 executable on thecomputer system 40 can be configured to control reading, displaying,delivering, receiving, assessing, evaluating and/or modifying bothacoustic and electric stimulation signals sent to the sound processingsystem 230. The fitting software 38 can generate a known acousticalsignal, which can be outputted through the sound source 130. The soundsource 130 can include one or more acoustical signal output devices suchas a speaker 132 or equivalent devices. In some implementations,multiple speakers 132 are positioned in a 2-D array to providedirectivity of the acoustical signal.

The computer system 40 executing the fitting software 38 can include adisplay screen for displaying selection screens, stimulation templatesand other information generated by the fitting software. The someimplementations, the computer system 40 includes a display device, astorage device, RAM, ROM, input/output (I/O) ports, a keyboard, and amouse. The display screen can be implemented to display a graphical userinterface (GUI) executed as a part of the software 38 includingselection screens, stimulation templates and other information generatedby the software 38. An audiologist, other medical personnel, or even theCI user can easily view and modify all information necessary to controla fitting process. In some implementations, the fitting portion 100 isincluded within the sound processing system 230 and can allow the CIuser to actively perform cochlear implant front end diagnostics.

In some implementations, the fitting portion 100 is implemented as astand alone system located at the office of the audiologist or othermedical personnel. The fitting portion 100 allows the audiologist orother medical personnel to customize a sound processing strategy for theCI user during an initial fitting process after the implantation of theCI. The CI user can return to the office for subsequent adjustments asneeded. The return visits may be required because the CI user may not befully aware of his/her sound processing needs initially, and the usermay need time to learn to discriminate between different sound signalsand become more perceptive of the sound quality provided by the soundprocessing strategy. The fitting system 110 is implemented to includeinterfaces using hardware, software, or a combination of both hardwareand software. For example, a simple set of hardware buttons, knobs,dials, slides, or similar interfaces can be implemented to select andadjust fitting parameters. The interfaces can also be implemented as aGUI displayed on a screen.

In some implementations, the fitting portion 100 is implemented as aportable system. The portable fitting system can be provided to the CIuser as an accessory device for allowing the CI user to adjust the soundprocessing strategy as needed. The initial fitting process may beperformed by the CI user aided by the audiologist or other medicalpersonnel. After the initial fitting process, the user may performsubsequent adjustments without having to visit the audiologist or othermedical personnel. The portable fitting system can be implemented toinclude simple user interfaces using hardware, software, or acombination of both hardware and software to facilitate the adjustmentprocess as described above for the stand alone system implementation.

FIGS. 3A-B show a functional block diagram of the auditory front endcustomization system 10. FIG. 3A depicts implementations using a singlemicrophone 312. The microphone 312 of the microphone system 210 detectsa known acoustical signal outputted by the sound source 130, and theacoustical signal is converted to an electrical signal by an acousticfront end (AFE) 322. The electrical signal is presented along at leastone signal path 311 of the sound processing system 230. The electricalsignal is converted to a digital signal by an analog to digitalconverter (A/D) 324. The digitized signal is amplified by an automaticgain control (AGC) 326 and delivered to the digital signal processor(DSP) 316 to generate appropriate digital stimulations to an array ofstimulating electrodes in a Micro Implantable Cochlear Stimulator (ICS)160.

In some implementations, multiple microphones are implemented asdepicted in FIG. 3B. The acoustical signals captured in each microphone312, 314 are communicated along separate signal paths 311, 315. Eachpath 311 and 315 respectively includes an acoustic front end (AFE1 322and AFE2 323), an analog to digital converter (A/D1 324 and A/D2 325),and an automatic gain control (AGC1 326 and AGC2 327). For example, in abeamforming implementation whereby two or more microphones may beimplemented to provide directivity of the sound, signals from theseparate signal paths 311, 315 are combined using a beamforming module329. The combined beamforming signal is processed by the DSP 316 togenerate the digital stimulations to be sent to the ICS 160.

In the implementations using multiple microphones, the microphone system210 includes two or more microphones 312, 314 positioned in multiplelocations. For example, in one implementation, the microphones 312, 314are implemented with one internal microphone positioned internallybehind the ear and one external microphone positioned near the pinnae.In an alternate implementation, a tube sound port is added to theinternal microphone to align the sound pickup location with the externalmicrophone. In yet another implementation, two internal microphones incoplanar alignment are positioned near the pinnae. In yet anotherimplementation still, two external microphones are positioned near thepinnae. Many other combinations of internal and external microphonesusing different number of microphones are possible.

FIG. 4 depicts a block diagram of the acoustical front end customizationsystem 10 including possible signal sampling locations 400, 410, and420. While only a single signal pathway 311 is shown, two or more signalpathways can be implemented to process signals captured through two ormore microphones. The generated known signal captured by the microphonesystem 210 is sampled at one or more locations 400, 410, and 420 alongthe signal pathway 311 of the sound processing system 230. The sampledsignal(s) can be received through the IU 140 and analyzed by the fittingsystem 110. The sampled signal(s) is/are compared with the knownacoustical signal generated by the fitting system 110 to determine anundesired spectral transformation of the sampled signal. The undesiredtransformation is dependent at least on the positioning of themicrophones, mismatched characteristics of the microphones, and physicalanatomy of the CI user's head and ear. The undesired transformation iseliminated by implementing one or more appropriate digital filters atthe corresponding sampling locations 400, 410, and 420 to filter out theundesired spectral transformation of the sampled signal.

The sampling locations 400, 410, and 420 in the signal pathway 311 canbe determined by the system 10 to include one or more locations afterthe ND converter 324. For example, the digitized signal (after the ND324) can be processed using one or more digital signal processors(DSPs). FIG. 4 shows two optional DSP1 328 and DSP2 330, but the totalnumber of DSPs implemented can vary based on the desired signalprocessing. DSP1 328 and DSP2 330 can be implemented, for example, as adigital filter to perform spectral modulation of the digital signal. Byproviding one or more sampling locations, the system 10 is capable ofadapting to individual signal processing schemes unique to each CI user.

FIG. 5 depicts a flowchart 500 of a process for implementing theauditory front end customization system 10. A known acoustical signal isgenerated and outputted by the fitting portion 100 at 505. The knownacoustical signal is received by the microphone system 210 at 510. At515, the received acoustical signal is transformed as an electricalsignal by the acoustic front end 322. Then at 520, the electrical signalis digitized via a ND 324. A decision can be made at 525 to sample thedigitized signal. If the decision is made to sample the signal, thesignal is processed for optimization at 540. The optimized signal canthen be forwarded to the AGC 326 at 555 before being sent to abeamforming module (now shown) or another DSP unit for generatingstimulus signals.

In some implementations, optimization of the sampled signal at 540 isperformed via the fitting system 110. Alternatively, in otherimplementations, the sound processing system 230 is implemented toperform the optimization by generating a DSP module within the soundprocessing system 230. In other implementations, the existing DSP module316 is configured to perform the optimization.

Optimizing the sampled electrical signal is accomplished through atleast four signal processing events. The electrical signal is sampledand a spectrum of the sampled signal is determined at 542. Thedetermined spectrum of the sampled signal is compared to the spectrum ofthe known acoustical signal to generate a ratio of the two spectra at544. The generated ratio represents the undesired transformation of thesampled signal due to the positioning of the microphones, mismatchedcharacteristics of the microphones, and physical anatomy of the user'shead and ear. The ratio generated is used as the basis for designing andgenerating a digital filter to eliminate the undesired transformation ofthe sampled signal at 546. The generated digital filter is placed at thecorresponding sampling locations 400, 410, and 420 to filter the sampledsignal at 548. The filtered signal is directed to the available signalprocessing unit on the signal path 311. The next available signalprocessing unit can vary depending on the signal processing schemedesigned for a particular CI user.

The transfer functions and the digital filter based on the transferfunctions generated through optimization at 540 can be implemented usingEquations 1 through 4 below.

$\begin{matrix}{{S({j\omega})} = {{F\left\lbrack {s(t)} \right\rbrack} = {\int_{- \infty}^{+ \infty}{{s(t)}^{{- {\omega}}\; t}{t}}}}} & (1) \\{{R({j\omega})} = {{F\left\lbrack {r(t)} \right\rbrack} = {\int_{- \infty}^{+ \infty}{{r(t)}^{{- {\omega}}\; t}{t}}}}} & (2) \\{{H({j\omega})} = \frac{R({j\omega})}{S({j\omega})}} & (3) \\{{G({j\omega})} = \frac{T({j\omega})}{H({j\omega})}} & (4)\end{matrix}$

The acoustic signal or stimulus generated from the sound source 130 iss(t) and has a corresponding Fourier transform S(jω). The signalcaptured or recorded from the microphone system 210 is r(t) and has acorresponding Fourier transform R(jω). The acoustical transfer functionfrom source to the microphone, H(jω), can then be characterized byEquation (3) above. If the target frequency response is specified byT(jω), then the compensation filter shape is given by Equation (4)above. This compensation filter is appropriately smoothed and then fitwith an realizable digital filter, which is then stored on the soundprocessing system 230 at the appropriate location(s). The digital filtercan be a finite-impulse-response (FIR) filter or aninfinite-impulse-response (IIR) filter. Any one of several standardmethods (see, e.g., Discrete Time Signal Processing, Oppenheim andSchafer, Prentice Hall (1989)) can be used to derive the digital filter.The entire sequence of operation just described can be performed by thefitting system 110.

In some implementations, processing events 542, 544, 546, and 548 areimplemented as a single processing event, combined as two processingevents or further subdivided into multiple processing events.

If the decision at 525 is not to sample the digital signal, the digitalsignal is forwarded to the next signal processing unit. For example, afirst optional digital signal processing (DSP1) can be presented at 530.At the conclusion of the first optional digital signal processing,another opportunity to sample the digital signal can be presented at535. A decision to sample the digital signal at 535 instructs thefitting system 110 to perform the signal optimization at 540. The signalprocessing events 542, 544, 546, 548 are carried out to filter out theundesired transformation and optimize the digital signal as describedabove. The optimized digital signal is then forwarded to the nextavailable signal processing unit on the signal path 311. For example,the AGC 326 can be provided at 555 to protect the optimized signalagainst an overdriven or underdriven signal and maintain adequatedemodulation signal amplitude while avoiding occasional noise spikes.

However, if the decision at 535 is not to sample the digital signal,then the digital signal can be forwarded directly to the AGC 326 andprocessed as described above at 555. Alternatively, another optionaldigital signal processing (DSP2) can be provided at 545. The gaincontrolled digital signal can be processed at 560 to allow for yetanother sampling opportunity. If the decision at 560 is to sample thegain controlled digital signal, the sampled gain controlled digitalsignal can be processed by the fitting system 110 to perform theoptimization at 540. The signal processing events 542, 544, and 546 arecarried out on the gain controlled digital signal to filter out theundesired transformation and optimize the gain controlled digital signalas described above. The gain controlled digital signal can then beprocessed by the DSP module 316 and forwarded to the stimulatingelectrode array 170 in the ICS 160 to provide appropriate auditory nervestimulations. If the decision at 560 is not to sample the gaincontrolled signal, then the signal can be directly forwarded to the DSPmodule 316 and the stimulating electrode array 170 in the ICS 160 asdescribed previously.

While FIG. 4 depicts three possible sampling locations 400, 410, and 420along the signal path 311, other sampling locations are within the scopeof this disclosure. The number and location of sampling locations canpartially depend on the combination of hardware and software elementsimplemented along the signal path 311 to perform a particular signalprocessing algorithm or scheme. For example, the number of optionaldigital signal processing units 328 and 320 can vary.

Further, in implementations whereby multiple microphones are utilized,the signal processing described in FIG. 5 can be implemented in parallelalong separate signal paths 311, 315 corresponding to each microphone312, 314 as described in FIGS. 3A-B. One implementation of the auditoryfront end customization system 10 utilizing multiple microphones is thebeamforming application. Beamforming provides directivity of theacoustical signal and allows the individual CI user to focus on adesired portion of the acoustical signal. In a noisy environment, theindividual CI user can focus on the speech of a certain speaker tofacilitate comprehension of such speech over confusing background noise.FIG. 6 depicts a functional block diagram of a beamforming strategy,which may be substantially as shown and described in a co-pending U.S.patent application Ser. No. 11/534,933, filed Sep. 25, 2006, which isincorporated herein by reference. The microphone system 210 can includemultiple microphones in various combinations of microphone types andlocations. In one implementation, as represented in FIG. 6, twomicrophones, MIC1 312 and MIC2 314, are utilized. Separate signal paths311 and 315 are provided for each microphone to process individualacoustical signal captured by MIC1 312 and MIC2 314. The soundprocessing system 230 coupled to MIC1 312 and MIC2 314 can besubstantially as shown in FIGS. 3A-B above. A separate signal processingsystem 230 can be coupled to each microphone or alternatively a singlesignal processing system 230 can be coupled to both microphones.

A known acoustical signal generated by the fitting portion 100 iscaptured by MIC1 312 and MIC2 314, and the captured acoustical signal isprocessed by the sound processing system 230 in two separate signalpaths 311, 315. The captured acoustical signal in signal path 311 isprocessed in parallel with the captured acoustical signal in signalpaths 315 according to the signal processing described in FIG. 5 above.After performing the signal processing, as described in FIG. 5 above, inparallel, the processed and optimized acoustical signal in each signalpath 311, 315 emerges as a processed digital signal. The processedsignals are combined as a single digital beamforming signal via an adder331. The combined beamforming signal is processed by the DSP module 316to generate a digital stimulating signal to stimulate the array ofstimulating electrodes 170 in the ICS 160 as describe above.

For the beamforming implementation, the optimization signal processingat 540 (in FIG. 5) to filter out the undesired transformation from thesampled signals is performed through optimization modules 640 and 650placed at any of the multiple locations along the signal paths 311 and315. In addition, the combined signal can be further optimized throughan optimization module 670 placed along the combined signal path 680 toadjust for any unwanted spectral transformation in the combinedbeamforming signal. The unwanted spectral transformation can bequantified by computing the spectrum of the signal 680, computing thespectrum of the known acoustical signal, and computing the ratio of thetwo spectra. The ratio, which is now the transfer-function of thecombined beamforming signal, can then be compared to a target frequencyresponse (as in 0042 and 0043) to create a custom optimization filter.

FIG. 7 depicts a general process 700 for implementing cochlear implantfront end customization. At 710, a known acoustic signal is generated.At 720, the generated acoustic signal is detected. The detected signalis processed along a signal path of the cochlear implant front end at730. At 740, the processed signal is sampled at one or more locationsalong the signal path. The sampled signal is compared against thegenerated known signal to determine a ratio at 750. A filter isgenerated based on the ratio to filter out an undesired transformationpresent in the sampled signal at 760.

In some implementations, the techniques for achieving beamforming asdescribed herein and depicted in FIGS. 1-7 may be implemented using oneor more computer programs comprising computer executable code stored ona computer readable medium and executing on the computer system 40, thesound processor portion 200, or the CI fitting portion 100, or allthree. The computer readable medium may include a hard disk drive, aflash memory device, a random access memory device such as DRAM andSDRAM, removable storage medium such as CD-ROM and DVD-ROM, a tape, afloppy disk, a CompactFlash memory card, a secure digital (SD) memorycard, or some other storage device. In some implementations, thecomputer executable code may include multiple portions or modules, witheach portion designed to perform a specific function described inconnection with FIGS. 1-7 above. In some implementations, the techniquesmay be implemented using hardware such as a microprocessor, amicrocontroller, an embedded microcontroller with internal memory, or anerasable programmable read only memory (EPROM) encoding computerexecutable instructions for performing the techniques described inconnection with FIGS. 1-7. In other implementations, the techniques maybe implemented using a combination of software and hardware.

Processors suitable for the execution of a computer program include, byway of example, both general and special purpose microprocessors, andany one or more processors of any kind of digital computer, includinggraphics processors, such as a GPU. Generally, the processor willreceive instructions and data from a read only memory or a random accessmemory or both. The essential elements of a computer are a processor forexecuting instructions and one or more memory devices for storinginstructions and data. Generally, a computer will also include, or beoperatively coupled to receive data from or transfer data to, or both,one or more mass storage devices for storing data, e.g., magnetic,magneto optical disks, or optical disks. Information carriers suitablefor embodying computer program instructions and data include all formsof non volatile memory, including by way of example semiconductor memorydevices, e.g., EPROM, EEPROM, and flash memory devices; magnetic disks,e.g., internal hard disks or removable disks; magneto optical disks; andCD ROM and DVD-ROM disks. The processor and the memory can besupplemented by, or incorporated in, special purpose logic circuitry.

To provide for interaction with a user, the systems and techniquesdescribed here can be implemented on a computer having a display device(e.g., a CRT (cathode ray tube) or LCD (liquid crystal display) monitor)for displaying information to the user and a keyboard and a pointingdevice (e.g., a mouse or a trackball) by which the user can provideinput to the computer. Other kinds of devices can be used to provide forinteraction with a user as well; for example, feedback provided to theuser can be any form of sensory feedback (e.g., visual feedback,auditory feedback, or tactile feedback); and input from the user can bereceived in any form, including acoustic, speech, or tactile input.

A number of implementations have been disclosed herein. Nevertheless, itwill be understood that various modifications may be made withoutdeparting from the scope of the disclosure. Accordingly, otherimplementations are within the scope of the following claims.

1. A method of optimizing audio signals received by a cochlear implant system, comprising: receiving a known audio signal at a non-implanted portion of the cochlear implant system; sampling the known audio signal at a location along a signal path to generate a sampled audio signal; comparing the sampled audio signal to the known audio signal to determine undesired transformation data; and using the undesired transformation data to modify a filter in the signal path.
 2. The method of claim 1, wherein the known audio signal is converted to a digital signal by an analog-to-digital circuit in the signal path before it is sampled.
 3. The method of claim 1, wherein determining the undesired transformation data comprises computing a ratio between the known audio signal and the sampled audio signal.
 4. The method of claim 1, wherein the known audio signal comprises a spectrum, the sampled audio signal comprises a spectrum, and the undesired transformation data comprises a ratio of the sampled audio signal spectrum and the known audio signal spectrum.
 5. The method of claim 1, wherein the known audio signal is represented by a Fourier transform, the sampled audio signal is represented by a Fourier transform, and the undesired transformation data comprises a ratio of the sampled audio signal Fourier transform and the known audio signal Fourier transform.
 6. The method of claim 1, wherein the filter comprises a digital filter.
 7. The method claim 1, wherein the signal path comprises an acoustic front end and an automatic gain control circuitry.
 8. The method of claim 1, wherein the known audio signal is received by at least one microphone.
 9. The method of claim 1, wherein the filter is modified to match a first response of a first microphone with a second response of a second microphone.
 10. The method of claim 1, wherein an output signal of the filter is configured for transmission to a cochlear implant.
 11. The method of claim 10, wherein the transmission is wireless.
 12. A system for optimizing audio signals received by a cochlear implant system, comprising: an external portion of a cochlear implant system configured to receive a known audio signal and to produce a captured audio signal; filtering circuitry in the external portion configured to produce a filtered output of the captured audio signal; and processing circuitry configured to compare the captured audio signal and the known audio signal and to produce undesired transformation data, the processing circuitry further configured to modify the filtering circuitry in accordance with the undesired transformation data.
 13. The system of claim 12, wherein the external portion comprises a behind the ear headpiece.
 14. The system of claim 12, further comprising a fitting system, wherein the processing circuitry comprises a portion of the fitting system.
 15. The system of claim 14, wherein the fitting system issues the known audio signal.
 16. The system of claim 15, wherein the fitting system issues the known audio signal from a speaker.
 17. The system of claim 12, wherein the filtered output is configured for transmission to a cochlear implant.
 18. The system of claim 17, further comprising gain control circuitry for processing the filtered output prior to transmission to the cochlear implant.
 19. The system of claim 12, further comprising analog-to-digital converter circuitry for digitizing the known audio signal and for providing the digitized known audio signal to the filtering circuitry.
 20. The system of claim 12, wherein the known audio signal is received by at least one microphone.
 21. The system of claim 20, wherein the filtering circuitry is modified to match a first response of a first microphone with a second response of a second microphone.
 22. The system of claim 12, wherein the processing circuitry is configured to produce the undesired transformation data by comparing a spectrum of the captured audio signal with a spectrum of the known audio signal.
 23. The system of claim 12, wherein the undesired transformation data comprises a ratio of a Fourier transform of the captured audio signal and a Fourier transform of the known audio signal. 